In certain applications it is desirable to convert an analog signal to digital form for processing as is done, for example, in presently commercially available digital audio disc players and in video equipment such as standards converters, frame synchronizers and time-base correctors. Also, television receivers have been proposed in which a substantial portion of the video processing, including interlace to non-interlace scan conversion (i.e., "progressive" scanning), is performed with digital rather than analog circuit elements. In such apparatus it is desirable to be able to impart delay to the signal in fractional increments of the sampling period for such purposes, for example, as correcting timing errors.
Once an analog signal has been converted to digital form, its value is known exactly only at the particular instants when it was sampled. In applications where it is desired to delay the signal by a fraction of the sampling interval, the usual approach is to "estimate" or interpolate the delayed signal from two or more adjacent samples of the input signal. FIG. 1 herein is exemplary of a known form of "two-point" linear interpolation filter 10 which uses a weighted sum of delayed and non-delayed signals to generate an estimate of a signal delayed by a fraction (K) of the signal sampling interval. As will be explained in detail subsequently, the delay of filter 10 varies as a function of the frequency of the signal to be delayed for delays other than those corresponding to integer multiples of one-half of the signal sampling interval. Moreover, the amplitude response of filter 10 undesirably varies as a function of frequency and the selected delay (K) of the filter.
The deficiencies in the amplitude and phase response of filter 10 tend to limit its usefulness to applications where the maximum input signal frequency is but a small fraction (e.g., one-eighth) of the sampling frequency. One solution to the problem would be to increase the sampling frequency but such a solution is not always practical in applications where the sampling frequency is fixed by other system design parameters, cost considerations or industry standards. This is the case, for example, in consumer products such as digital television receivers, digital audio disc players or the like.
Another possible solution would be to provide the desired delay by means of a "higher-order" interpolator which utilizes more samples of the input signal in forming the delayed signal. Quadratic interpolators, for example, form a weighted sum of four samples of the input signal in generating a delayed signal and have superior amplitude and phase response as compared with two-point linear interpolators. As to be expected, however, the improved performance of higher-order interpolators is obtained at the cost of substantially increased circuit complexity particularly with regard to the number of complex digital arithmetic operations which must be performed such as multiplication by variables and additions of numbers.